Abstract
An analytical model is presented for computing the end-to-end voice call performance in a packet network which drops the less significant bits in voice packets during periods of congestion. The authors first modify an existing single-node bit-dropping model to include the situation when the overall arrival process at a node consists of a mix of packets of different sizes. Variability in packet size is caused by bit dropping at previous nodes. A detailed model is presented to capture bit dropping effects in a tandem connection of nodes. The model includes the effects of load fluctuations at each node, and also takes into account the dependencies in bit dropping experienced by a voice packet at successive nodes in a tandem connection. These dependencies occur because a packet that has experienced bit dropping at a node in congestion state Q would not be further bit-dropped at a subsequent node if its current congestion state is no worse than Q. The model also incorporates the reduction in packet service times at intermediate nodes due to dropping at previous nodes. In addition to the detailed model, the authors present two approximation procedures which also serve as upper and lower bounds. In particular, the upper bound is shown to be very tight for practical range of loads, and hence serves as a good approximation with significant computational simplicity.

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